Traditional circuit-switched telephone networks and packet-switched data networks are converging to an increasing extent. One reason for this is that by combining both types of network into just one network it is possible to save on installation, maintenance and expansion costs, since the latter no longer accrue for two networks, but henceforth only for one. Another reason is that more and more voice communication takes place on a packet-switched basis already on the subscriber side (“Voice over IP”: “VoIP” for short). Consequently this technology lends itself to use at network level.
Considerably less data is transferred for voice communication in packet-switched networks, which in itself is already equivalent to a substantial potential gain in efficiency. This is achieved by means of different compression methods (referred to as “codecs”, standing for “coder-decoder”). The most diverse types of codecs are used for data compression. The G.711 codec operates e.g. at a sampling rate of 64 kHz, while the G.729 codec, for example, has a sampling rate of only 8 kHz.
Different protocols are also used. An example hereof is the H.323 protocol which was created by the ITU. This protocol in turn comprises various subprotocols that are used for different applications, such as e.g. H.261 for video, G.711 for audio, T.122 for data, and H.225 for signaling. An alternative to the H.323 protocol is the so-called Session Initiation Protocol (SIP).
Whereas certain applications in the network domain, e.g. Video on Demand, place no particular real-time requirements on the network, voice communication imposes high real-time requirements, since even brief delays are perceived by people as annoying during a telephone conversation.
Various approaches have been developed in order to enable high quality in particular for voice connections, all of which approaches focus on the quality of service (“QoS”). These include for example:                possible ways of reducing data congestion, or avoiding occurrences thereof from the outset (“congestion management”),        handling data traffic in accordance with what are termed “service levels”; in this scheme a distinction is made between different service levels for the data traffic; different queues are set up, e.g. in routers, for the respective service levels and processed accordingly (“classification and queuing techniques”),        use of identifiers for packets, which identifiers specify a particular path of the packet through the network along network nodes (“packet tagging/label switching”)        provisioning of bandwidth in all domains of the network, both in bandwidth-critical as well as in non-bandwidth-critical domains of the network (“overprovisioning”).        
A continuous, end-to-end method for managing bandwidth across different domains, in order to guarantee the availability of a requested bandwidth for real-time-critical connections in particular, is not known in packet-oriented communication networks according to the prior art.
The TIPHON standard (“Telecommunications and Internet Protocol Harmonization over Networks”) developed by ETSI (“European Telecommunications Standards Institute”) has so far not succeeded in establishing itself, not least due to its complexity.
Other prior art approaches to solving the aforesaid problem provide without exception island solutions which do not work effectively with meshed network structures that are characterized e.g. by different administrative domains. A request to reserve the bandwidth must be made each time a connection is set up. In the island solutions this is always a reservation request to the network-wide bandwidth management function. As a result all requests must be carried throughout the entire network in each case. Even requests which actually require no reservation are conveyed across the entire network.